similar to: Polycom VSX 7000e Series & Asterisk

Hello, Has anyone used Polycom's VSX line of videoconferencing equipment with Asterisk? It seems some of their models, namely the newer VSX 5000, supports SIP. -- The Internet used to be a lot of smart people sitting at dumb terminals, but now its a lot of dumb people sitting at smart terminals!

Author: aj Repository: /hg/zfs-crypto/gate Revision: 7567fbd5fb7e1615c352bd1e1f9cdeba24d5e961 Log message: 6406622 UNIX98/UNIX03: *vsx* Trusted Extensions changes to tar.h pollute standards namespace Files: update: usr/src/head/tar.h

Author: casper Repository: /hg/zfs-crypto/gate Revision: 24515d11f294677252853b5e1eb496c66b1e06e8 Log message: 6395215 UNIX03/UNIX98: *vsx* addition of mkstemps breaks namespace Files: update: usr/src/head/stdlib.h

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

I bought a couple Polycom Soundpoint 300's, and have them working nicely with SIP... but I'd like to be able to do automatic config via FTP, but it requires some XML config files. The docs discuss them in detail, but I can't seem to d/l them from Polycom. [No, it doesn't appear to be on the CD that comes with the phone.] I've created an account at Polycom's tech support

Ken - take a look at using IAX protocol to route calls between your Asterisk boxes. Cory Andrews -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ken Williams Sent: Wednesday, November 01, 2006 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM HEADER] - RE: [asterisk-users]

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between "split" and "combined" firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I don't quite see how that works. Do you point a non-VLAN'd segment at it (akin to when you uplink a VLAN_enabled switch), and have the phone implement the VLAN? Or...? *puzzled* Thanks much, -Ken

I was wondering if anyone can help me with a problem we have at one of our sites. We have setup a Asterisk Trunk to a Avaya PBX, ie ... Avaya <-> Asterisk (1.2.30) <-> External ISDN Network BUT They also have a Polycom VSX 7000 that with some sort of BRI converters that plugs into the Avaya. The Trunk is working well except for Video Conference Calls. The Polocom can receive

Hello everyone, I have just received 3 brand new Polycom SoundPoint IP 600 from "voipsupply.com" and I have the exact same problem on all of them. When I receive a call, the phone is ringing correctly but when I answer it, it takes exactly 10 seconds before I can hear the caller. I also have SoundPoint 300 and 301 and I don't have that problem with those. I'm using Asterisk

I've had phones before where, with the phone on-hook, it still implements the local dialplan. E.g., if I dialed "0" (on-hook), after three seconds, it would dial the operator, and have the call on speakerphone. Does Polycom allow this functionality? Clearly, not a necessary feature... but it would be a nice one. Thanks! -Ken -- This message has been scanned for viruses and

Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble on registration(? -- maybe it's on acquiring an IP?) has started again. I still have the old sip.cfg, but can't figure out which option it is. Any help? Thanks! -Ken D'Ambrosio -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hey, all. I've got an office set up with Asterisk, and forwarding's got a bit of a glitch: When they forward, they listen for the remote phone to ring, then hang up. If the remote phone doesn't connect, it goes to the original phone's VM. Is this Polycom's "fault," or Asterisk's? I've been reading up on blind/supervised forwards, and, honestly, have

Hey, all. I'm all over MWI, but I gotta say that I think the Polycoms go a bit over the top. The blinking LED is enough for me; how do I disable the stuttered dialtone and the audible warble? I've looked through the config files, but there are a HELL of a lot of options, and I haven't been able to find those particular ones yet. Thanks! -Ken -- This message has been scanned for

I can't figure out how one upgrades the boot ROMs for the Polycoms. I've read the wiki, I've checked the docs, I've looked here, there, everywhere; I hoped it might be through the web interface, through DHCP... but I can't see how to do it. Suggestions? Thanks! -Ken

Came upon a problem today that I thought I'd see if it's by design, if I'm missing an option somewhere, or if my fix is the way to fix it. We setup a remote location with a server, same as we've done with others, but for some reason when they would transfer an outside call anywhere it would pause for a few seconds and hang up the line. Well, after spending most of the day on

This is a SER/Polycom question, but I hoped we may have some SER guru's here... I have a series of Polycom phones that are tying to register with OpenSER. The phone sends a REGISTER message, and OpenSER replies with Unauthorised (all normal). The phone re-sends the REGISTER with the credentials, and OpenSER sends Ok. Here's where it goes downhill. The polycom's appearance display

Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio

Hey, all. I'm working on making a script to auto-provision my Polycoms. I wanted one that: - Gets the MAC by itself - Fills in the provisioning info you supplied on a web page - Creates appropriate files - Reboots the phone (which then gets provisioned) The last part was the sticking one, though. I found plenty of ways to make them reboot -- but most required an already-provisioned phone,